Network voice and video communication systems and applications, such as Voice over Internet Protocol (VoIP) systems, Skype®, or Skype® for Business systems, have become popular platforms for not only providing voice calls between users, but also for video calls, live meeting hosting, interactive white boarding, and other point-to-point or multi-user network-based communications. These network telephony systems typically rely upon packet communications and packet routing, such as the Internet, instead of traditional circuit-switched communications, such as the Public Switched Telephone Network (PSTN) or circuit-switched cellular networks.
In many examples, communication links can be established among one or more endpoints, such as user devices, to provide voice and video calls or interactive conferencing within specialized software applications on computers, laptops, tablet devices, smartphones, gaming systems, and the like. As these network telephony systems have grown in popularity, associated traffic volumes have increased and efficient use of network resources that carry this traffic has been difficult to achieve. Among these difficulties is efficient encoding and decoding of speech content for transfer among endpoints, as well as reducing lag or latency in speech exchanged among endpoints due to network delays and encoding/decoding delays. Although various high-compression audio and video encoding/decoding algorithms (codecs) have been developed over the years, these codecs can still produce undesirable voice or speech lag among endpoints.